This invention relates generally to call signaling and, more specifically, however, to using multicast call signaling in a packet network.
Conducting voice communications over a packet network is desirable because companies typically use 50% or less of available network bandwidth. Therefore, putting company voice traffic on the same packet network used to transmit data can save long distance charges and allow for greater communication options. Sending voice data on IP networks is therefore extremely desirable when there is a guaranteed bandwidth.
Call signaling techniques in packet networks traditionally route signaling data serially, or sequentially, such that a response is sought from only one endpoint at a time. If a first endpoint fails to respond to the request, then the request is sent to a second endpoint. If no response is obtained from the second endpoint the request is sent to a third endpoint, and so on. A conventional telephony call forwarding technique is used to connect with a user at any one of multiple locations—such as attempting to contact someone whether they are at the office by an office phone, at home by a home phone, or away from home and the office by a cell phone or a pager—operates as follows. First, the telephony system attempts to reach the desired party at the office by signaling (calling) the office phone. If no response is obtained at the office, the system calls the home phone. If no response is received from the home phone, then the cell phone is signaled. Finally, the pager is contacted if no response was obtained from any of the previous devices. When a response is finally obtained, either from a live-person contact, from an answering machine, or from any other type of coverage device (i.e., voice mail), the signaling sequence stops and no further devices are signaled.
Unfortunately, not only does this system consume a great deal of time trying to track down the desired party, it also breaks down when there is an answering machine or voice messaging device at one of the endpoints. When the answering machine or voice mail system establishes a connection with the caller, the call forwarding process stops and the desired contact must be obtained by calling each phone number manually.
Another problem related to sequential call signaling involves conference calling. Conference calls generally require that each party be contacted separately. Specifically, one party is typically called and put on hold while the next party is contacted. This process continues until all of the desired parties are finally contacted. Again, this method is time consuming and burdensome. Scalability is also a significant problem because all of the work done in contacting the desired parties is done at the initiating endpoint.
The existing audio transport protocol used in IP networks H.323 is derived from traditional telephone networks and is not highly scalable in packet networks or even Local Area Network (LAN) environments. Telephony based protocols are also complex and bulky and require centralized control. Thus, these traditional audio transport protocols are not easily implemented in low cost devices such as cable modems, phones, and pagers because of the intensive processing requirements.
Therefore, a need remains for a scalable and easily implemented packet-based call signaling system that contacts multiple communications devices at the same time. The industry would also benefit from a call signaling system that allows multiple parties to be contacted for participation in a conference call, without the need to separately call each desired party.